14b43beb7c
TBR=niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/915006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@2963 4adac7df-926f-26a2-2b94-8c16560cd09d
947 lines
27 KiB
C++
947 lines
27 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_H_
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#define WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_H_
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#include <vector>
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#include "modules/interface/module.h"
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#include "modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
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namespace webrtc {
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// forward declaration
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class RemoteBitrateEstimator;
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class RemoteBitrateObserver;
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class Transport;
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class RtpRtcp : public Module {
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public:
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struct Configuration {
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Configuration()
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: id(-1),
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audio(false),
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clock(NULL),
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default_module(NULL),
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incoming_data(NULL),
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incoming_messages(NULL),
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outgoing_transport(NULL),
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rtcp_feedback(NULL),
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intra_frame_callback(NULL),
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bandwidth_callback(NULL),
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audio_messages(NULL),
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remote_bitrate_estimator(NULL) {
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}
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/* id - Unique identifier of this RTP/RTCP module object
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* audio - True for a audio version of the RTP/RTCP module
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* object false will create a video version
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* clock - The clock to use to read time. If NULL object
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* will be using the system clock.
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* incoming_data - Callback object that will receive the incoming
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* data
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* incoming_messages - Callback object that will receive the incoming
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* RTP messages.
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* outgoing_transport - Transport object that will be called when packets
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* are ready to be sent out on the network
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* rtcp_feedback - Callback object that will receive the incoming
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* RTP messages.
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* intra_frame_callback - Called when the receiver request a intra frame.
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* bandwidth_callback - Called when we receive a changed estimate from
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* the receiver of out stream.
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* audio_messages - Telehone events.
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* remote_bitrate_estimator - Estimates the bandwidth available for a set of
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* streams from the same client.
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*/
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int32_t id;
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bool audio;
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RtpRtcpClock* clock;
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RtpRtcp* default_module;
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RtpData* incoming_data;
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RtpFeedback* incoming_messages;
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Transport* outgoing_transport;
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RtcpFeedback* rtcp_feedback;
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RtcpIntraFrameObserver* intra_frame_callback;
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RtcpBandwidthObserver* bandwidth_callback;
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RtpAudioFeedback* audio_messages;
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RemoteBitrateEstimator* remote_bitrate_estimator;
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};
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/*
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* Create a RTP/RTCP module object using the system clock.
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*
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* configuration - Configuration of the RTP/RTCP module.
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*/
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static RtpRtcp* CreateRtpRtcp(const RtpRtcp::Configuration& configuration);
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/**************************************************************************
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*
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* Receiver functions
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*
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***************************************************************************/
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/*
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* configure a RTP packet timeout value
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*
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* RTPtimeoutMS - time in milliseconds after last received RTP packet
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* RTCPtimeoutMS - time in milliseconds after last received RTCP packet
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*
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* return -1 on failure else 0
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*/
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virtual WebRtc_Word32 SetPacketTimeout(
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const WebRtc_UWord32 RTPtimeoutMS,
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const WebRtc_UWord32 RTCPtimeoutMS) = 0;
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/*
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* Set periodic dead or alive notification
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*
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* enable - turn periodic dead or alive notification on/off
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* sampleTimeSeconds - sample interval in seconds for dead or alive
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* notifications
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*
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* return -1 on failure else 0
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*/
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virtual WebRtc_Word32 SetPeriodicDeadOrAliveStatus(
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const bool enable,
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const WebRtc_UWord8 sampleTimeSeconds) = 0;
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/*
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* Get periodic dead or alive notification status
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*
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* enable - periodic dead or alive notification on/off
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* sampleTimeSeconds - sample interval in seconds for dead or alive
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* notifications
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*
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* return -1 on failure else 0
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*/
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virtual WebRtc_Word32 PeriodicDeadOrAliveStatus(
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bool& enable,
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WebRtc_UWord8& sampleTimeSeconds) = 0;
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/*
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* set voice codec name and payload type
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*
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* return -1 on failure else 0
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*/
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virtual WebRtc_Word32 RegisterReceivePayload(
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const CodecInst& voiceCodec) = 0;
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/*
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* set video codec name and payload type
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*
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* return -1 on failure else 0
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*/
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virtual WebRtc_Word32 RegisterReceivePayload(
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const VideoCodec& videoCodec) = 0;
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/*
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* get payload type for a voice codec
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*
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* return -1 on failure else 0
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*/
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virtual WebRtc_Word32 ReceivePayloadType(
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const CodecInst& voiceCodec,
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WebRtc_Word8* plType) = 0;
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/*
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* get payload type for a video codec
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*
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* return -1 on failure else 0
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*/
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virtual WebRtc_Word32 ReceivePayloadType(
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const VideoCodec& videoCodec,
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WebRtc_Word8* plType) = 0;
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/*
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* Remove a registered payload type from list of accepted payloads
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*
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* payloadType - payload type of codec
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*
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* return -1 on failure else 0
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*/
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virtual WebRtc_Word32 DeRegisterReceivePayload(
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const WebRtc_Word8 payloadType) = 0;
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/*
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* (De)register RTP header extension type and id.
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*
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* return -1 on failure else 0
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*/
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virtual WebRtc_Word32 RegisterReceiveRtpHeaderExtension(
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const RTPExtensionType type,
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const WebRtc_UWord8 id) = 0;
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virtual WebRtc_Word32 DeregisterReceiveRtpHeaderExtension(
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const RTPExtensionType type) = 0;
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/*
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* Get last received remote timestamp
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*/
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virtual WebRtc_UWord32 RemoteTimestamp() const = 0;
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/*
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* Get the local time of the last received remote timestamp
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*/
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virtual int64_t LocalTimeOfRemoteTimeStamp() const = 0;
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/*
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* Get the current estimated remote timestamp
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*
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* timestamp - estimated timestamp
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*
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* return -1 on failure else 0
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*/
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virtual WebRtc_Word32 EstimatedRemoteTimeStamp(
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WebRtc_UWord32& timestamp) const = 0;
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/*
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* Get incoming SSRC
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*/
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virtual WebRtc_UWord32 RemoteSSRC() const = 0;
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/*
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* Get remote CSRC
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*
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* arrOfCSRC - array that will receive the CSRCs
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*
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* return -1 on failure else the number of valid entries in the list
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*/
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virtual WebRtc_Word32 RemoteCSRCs(
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WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize]) const = 0;
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/*
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* get the currently configured SSRC filter
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*
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* allowedSSRC - SSRC that will be allowed through
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*
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* return -1 on failure else 0
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*/
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virtual WebRtc_Word32 SSRCFilter(WebRtc_UWord32& allowedSSRC) const = 0;
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/*
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* set a SSRC to be used as a filter for incoming RTP streams
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*
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* allowedSSRC - SSRC that will be allowed through
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*
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* return -1 on failure else 0
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*/
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virtual WebRtc_Word32 SetSSRCFilter(const bool enable,
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const WebRtc_UWord32 allowedSSRC) = 0;
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/*
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* Turn on/off receiving RTX (RFC 4588) on a specific SSRC.
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*/
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virtual WebRtc_Word32 SetRTXReceiveStatus(const bool enable,
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const WebRtc_UWord32 SSRC) = 0;
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/*
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* Get status of receiving RTX (RFC 4588) on a specific SSRC.
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*/
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virtual WebRtc_Word32 RTXReceiveStatus(bool* enable,
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WebRtc_UWord32* SSRC) const = 0;
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/*
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* called by the network module when we receive a packet
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*
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* incomingPacket - incoming packet buffer
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* packetLength - length of incoming buffer
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*
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* return -1 on failure else 0
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*/
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virtual WebRtc_Word32 IncomingPacket(const WebRtc_UWord8* incomingPacket,
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const WebRtc_UWord16 packetLength) = 0;
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/**************************************************************************
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*
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* Sender
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*
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***************************************************************************/
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/*
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* set MTU
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*
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* size - Max transfer unit in bytes, default is 1500
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*
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* return -1 on failure else 0
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*/
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virtual WebRtc_Word32 SetMaxTransferUnit(const WebRtc_UWord16 size) = 0;
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/*
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* set transtport overhead
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* default is IPv4 and UDP with no encryption
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*
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* TCP - true for TCP false UDP
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* IPv6 - true for IP version 6 false for version 4
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* authenticationOverhead - number of bytes to leave for an
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* authentication header
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*
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* return -1 on failure else 0
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*/
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virtual WebRtc_Word32 SetTransportOverhead(
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const bool TCP,
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const bool IPV6,
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const WebRtc_UWord8 authenticationOverhead = 0) = 0;
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/*
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* Get max payload length
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*
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* A combination of the configuration MaxTransferUnit and
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* TransportOverhead.
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* Does not account FEC/ULP/RED overhead if FEC is enabled.
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* Does not account for RTP headers
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*/
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virtual WebRtc_UWord16 MaxPayloadLength() const = 0;
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/*
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* Get max data payload length
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*
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* A combination of the configuration MaxTransferUnit, headers and
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* TransportOverhead.
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* Takes into account FEC/ULP/RED overhead if FEC is enabled.
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* Takes into account RTP headers
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*/
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virtual WebRtc_UWord16 MaxDataPayloadLength() const = 0;
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/*
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* set codec name and payload type
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*
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* return -1 on failure else 0
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*/
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virtual WebRtc_Word32 RegisterSendPayload(
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const CodecInst& voiceCodec) = 0;
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/*
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* set codec name and payload type
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*
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* return -1 on failure else 0
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*/
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virtual WebRtc_Word32 RegisterSendPayload(
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const VideoCodec& videoCodec) = 0;
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/*
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* Unregister a send payload
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*
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* payloadType - payload type of codec
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*
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* return -1 on failure else 0
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*/
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virtual WebRtc_Word32 DeRegisterSendPayload(
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const WebRtc_Word8 payloadType) = 0;
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/*
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* (De)register RTP header extension type and id.
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*
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* return -1 on failure else 0
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*/
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virtual WebRtc_Word32 RegisterSendRtpHeaderExtension(
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const RTPExtensionType type,
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const WebRtc_UWord8 id) = 0;
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virtual WebRtc_Word32 DeregisterSendRtpHeaderExtension(
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const RTPExtensionType type) = 0;
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/*
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* Enable/disable traffic smoothing of sending stream.
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*/
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virtual void SetTransmissionSmoothingStatus(const bool enable) = 0;
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virtual bool TransmissionSmoothingStatus() const = 0;
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/*
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* get start timestamp
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*/
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virtual WebRtc_UWord32 StartTimestamp() const = 0;
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/*
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* configure start timestamp, default is a random number
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*
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* timestamp - start timestamp
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*
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* return -1 on failure else 0
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*/
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virtual WebRtc_Word32 SetStartTimestamp(
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const WebRtc_UWord32 timestamp) = 0;
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/*
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* Get SequenceNumber
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*/
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virtual WebRtc_UWord16 SequenceNumber() const = 0;
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/*
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* Set SequenceNumber, default is a random number
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*
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* return -1 on failure else 0
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*/
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virtual WebRtc_Word32 SetSequenceNumber(const WebRtc_UWord16 seq) = 0;
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/*
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* Get SSRC
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*/
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virtual WebRtc_UWord32 SSRC() const = 0;
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/*
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* configure SSRC, default is a random number
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*
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* return -1 on failure else 0
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*/
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virtual WebRtc_Word32 SetSSRC(const WebRtc_UWord32 ssrc) = 0;
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/*
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* Get CSRC
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*
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* arrOfCSRC - array of CSRCs
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*
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* return -1 on failure else number of valid entries in the array
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*/
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virtual WebRtc_Word32 CSRCs(
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WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize]) const = 0;
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/*
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* Set CSRC
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*
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* arrOfCSRC - array of CSRCs
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* arrLength - number of valid entries in the array
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*
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* return -1 on failure else 0
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*/
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virtual WebRtc_Word32 SetCSRCs(
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const WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize],
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const WebRtc_UWord8 arrLength) = 0;
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/*
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* includes CSRCs in RTP header if enabled
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*
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* include CSRC - on/off
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*
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* default:on
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*
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* return -1 on failure else 0
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*/
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virtual WebRtc_Word32 SetCSRCStatus(const bool include) = 0;
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/*
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* Turn on/off sending RTX (RFC 4588) on a specific SSRC.
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*/
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virtual WebRtc_Word32 SetRTXSendStatus(const bool enable,
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const bool setSSRC,
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const WebRtc_UWord32 SSRC) = 0;
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/*
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* Get status of sending RTX (RFC 4588) on a specific SSRC.
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*/
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virtual WebRtc_Word32 RTXSendStatus(bool* enable,
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WebRtc_UWord32* SSRC) const = 0;
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/*
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* sends kRtcpByeCode when going from true to false
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*
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* sending - on/off
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*
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* return -1 on failure else 0
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*/
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virtual WebRtc_Word32 SetSendingStatus(const bool sending) = 0;
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/*
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* get send status
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*/
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virtual bool Sending() const = 0;
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/*
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* Starts/Stops media packets, on by default
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*
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* sending - on/off
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*
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* return -1 on failure else 0
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*/
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virtual WebRtc_Word32 SetSendingMediaStatus(const bool sending) = 0;
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/*
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* get send status
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*/
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virtual bool SendingMedia() const = 0;
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/*
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* get sent bitrate in Kbit/s
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*/
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virtual void BitrateSent(WebRtc_UWord32* totalRate,
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WebRtc_UWord32* videoRate,
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WebRtc_UWord32* fecRate,
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WebRtc_UWord32* nackRate) const = 0;
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/*
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* Get the receive-side estimate of the available bandwidth.
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*/
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virtual int EstimatedReceiveBandwidth(
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WebRtc_UWord32* available_bandwidth) const = 0;
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/*
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* Used by the codec module to deliver a video or audio frame for
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* packetization.
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*
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* frameType - type of frame to send
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* payloadType - payload type of frame to send
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* timestamp - timestamp of frame to send
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* payloadData - payload buffer of frame to send
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* payloadSize - size of payload buffer to send
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* fragmentation - fragmentation offset data for fragmented frames such
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* as layers or RED
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*
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* return -1 on failure else 0
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*/
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virtual WebRtc_Word32 SendOutgoingData(
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const FrameType frameType,
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const WebRtc_Word8 payloadType,
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const WebRtc_UWord32 timeStamp,
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int64_t capture_time_ms,
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const WebRtc_UWord8* payloadData,
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const WebRtc_UWord32 payloadSize,
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const RTPFragmentationHeader* fragmentation = NULL,
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const RTPVideoHeader* rtpVideoHdr = NULL) = 0;
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/**************************************************************************
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*
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* RTCP
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*
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***************************************************************************/
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/*
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* Get RTCP status
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*/
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virtual RTCPMethod RTCP() const = 0;
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/*
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* configure RTCP status i.e on(compound or non- compound)/off
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*
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* method - RTCP method to use
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*
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* return -1 on failure else 0
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*/
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virtual WebRtc_Word32 SetRTCPStatus(const RTCPMethod method) = 0;
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/*
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* Set RTCP CName (i.e unique identifier)
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*
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* return -1 on failure else 0
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*/
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virtual WebRtc_Word32 SetCNAME(const char cName[RTCP_CNAME_SIZE]) = 0;
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/*
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* Get RTCP CName (i.e unique identifier)
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*
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* return -1 on failure else 0
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*/
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virtual WebRtc_Word32 CNAME(char cName[RTCP_CNAME_SIZE]) = 0;
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/*
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* Get remote CName
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*
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* return -1 on failure else 0
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*/
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virtual WebRtc_Word32 RemoteCNAME(
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const WebRtc_UWord32 remoteSSRC,
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char cName[RTCP_CNAME_SIZE]) const = 0;
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/*
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* Get remote NTP
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*
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* return -1 on failure else 0
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*/
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virtual WebRtc_Word32 RemoteNTP(
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WebRtc_UWord32 *ReceivedNTPsecs,
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WebRtc_UWord32 *ReceivedNTPfrac,
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WebRtc_UWord32 *RTCPArrivalTimeSecs,
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WebRtc_UWord32 *RTCPArrivalTimeFrac,
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WebRtc_UWord32 *rtcp_timestamp) const = 0;
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/*
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* AddMixedCNAME
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*
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* return -1 on failure else 0
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*/
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virtual WebRtc_Word32 AddMixedCNAME(
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|
const WebRtc_UWord32 SSRC,
|
|
const char cName[RTCP_CNAME_SIZE]) = 0;
|
|
|
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/*
|
|
* RemoveMixedCNAME
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
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virtual WebRtc_Word32 RemoveMixedCNAME(const WebRtc_UWord32 SSRC) = 0;
|
|
|
|
/*
|
|
* Get RoundTripTime
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual WebRtc_Word32 RTT(const WebRtc_UWord32 remoteSSRC,
|
|
WebRtc_UWord16* RTT,
|
|
WebRtc_UWord16* avgRTT,
|
|
WebRtc_UWord16* minRTT,
|
|
WebRtc_UWord16* maxRTT) const = 0 ;
|
|
|
|
/*
|
|
* Reset RoundTripTime statistics
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual WebRtc_Word32 ResetRTT(const WebRtc_UWord32 remoteSSRC)= 0 ;
|
|
|
|
/*
|
|
* Force a send of a RTCP packet
|
|
* normal SR and RR are triggered via the process function
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual WebRtc_Word32 SendRTCP(
|
|
WebRtc_UWord32 rtcpPacketType = kRtcpReport) = 0;
|
|
|
|
/*
|
|
* Good state of RTP receiver inform sender
|
|
*/
|
|
virtual WebRtc_Word32 SendRTCPReferencePictureSelection(
|
|
const WebRtc_UWord64 pictureID) = 0;
|
|
|
|
/*
|
|
* Send a RTCP Slice Loss Indication (SLI)
|
|
* 6 least significant bits of pictureID
|
|
*/
|
|
virtual WebRtc_Word32 SendRTCPSliceLossIndication(
|
|
const WebRtc_UWord8 pictureID) = 0;
|
|
|
|
/*
|
|
* Reset RTP statistics
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual WebRtc_Word32 ResetStatisticsRTP() = 0;
|
|
|
|
/*
|
|
* statistics of our localy created statistics of the received RTP stream
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual WebRtc_Word32 StatisticsRTP(
|
|
WebRtc_UWord8* fraction_lost, // scale 0 to 255
|
|
WebRtc_UWord32* cum_lost, // number of lost packets
|
|
WebRtc_UWord32* ext_max, // highest sequence number received
|
|
WebRtc_UWord32* jitter,
|
|
WebRtc_UWord32* max_jitter = NULL) const = 0;
|
|
|
|
/*
|
|
* Reset RTP data counters for the receiving side
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual WebRtc_Word32 ResetReceiveDataCountersRTP() = 0;
|
|
|
|
/*
|
|
* Reset RTP data counters for the sending side
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual WebRtc_Word32 ResetSendDataCountersRTP() = 0;
|
|
|
|
/*
|
|
* statistics of the amount of data sent and received
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual WebRtc_Word32 DataCountersRTP(
|
|
WebRtc_UWord32* bytesSent,
|
|
WebRtc_UWord32* packetsSent,
|
|
WebRtc_UWord32* bytesReceived,
|
|
WebRtc_UWord32* packetsReceived) const = 0;
|
|
/*
|
|
* Get received RTCP sender info
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual WebRtc_Word32 RemoteRTCPStat(RTCPSenderInfo* senderInfo) = 0;
|
|
|
|
/*
|
|
* Get received RTCP report block
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual WebRtc_Word32 RemoteRTCPStat(
|
|
std::vector<RTCPReportBlock>* receiveBlocks) const = 0;
|
|
/*
|
|
* Set received RTCP report block
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual WebRtc_Word32 AddRTCPReportBlock(
|
|
const WebRtc_UWord32 SSRC,
|
|
const RTCPReportBlock* receiveBlock) = 0;
|
|
|
|
/*
|
|
* RemoveRTCPReportBlock
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual WebRtc_Word32 RemoveRTCPReportBlock(const WebRtc_UWord32 SSRC) = 0;
|
|
|
|
/*
|
|
* (APP) Application specific data
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual WebRtc_Word32 SetRTCPApplicationSpecificData(
|
|
const WebRtc_UWord8 subType,
|
|
const WebRtc_UWord32 name,
|
|
const WebRtc_UWord8* data,
|
|
const WebRtc_UWord16 length) = 0;
|
|
/*
|
|
* (XR) VOIP metric
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual WebRtc_Word32 SetRTCPVoIPMetrics(
|
|
const RTCPVoIPMetric* VoIPMetric) = 0;
|
|
|
|
/*
|
|
* (REMB) Receiver Estimated Max Bitrate
|
|
*/
|
|
virtual bool REMB() const = 0;
|
|
|
|
virtual WebRtc_Word32 SetREMBStatus(const bool enable) = 0;
|
|
|
|
virtual WebRtc_Word32 SetREMBData(const WebRtc_UWord32 bitrate,
|
|
const WebRtc_UWord8 numberOfSSRC,
|
|
const WebRtc_UWord32* SSRC) = 0;
|
|
|
|
/*
|
|
* (IJ) Extended jitter report.
|
|
*/
|
|
virtual bool IJ() const = 0;
|
|
|
|
virtual WebRtc_Word32 SetIJStatus(const bool enable) = 0;
|
|
|
|
/*
|
|
* (TMMBR) Temporary Max Media Bit Rate
|
|
*/
|
|
virtual bool TMMBR() const = 0;
|
|
|
|
/*
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual WebRtc_Word32 SetTMMBRStatus(const bool enable) = 0;
|
|
|
|
/*
|
|
* (NACK)
|
|
*/
|
|
virtual NACKMethod NACK() const = 0;
|
|
|
|
/*
|
|
* Turn negative acknowledgement requests on/off
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual WebRtc_Word32 SetNACKStatus(const NACKMethod method) = 0;
|
|
|
|
/*
|
|
* TODO(holmer): Propagate this API to VideoEngine.
|
|
* Returns the currently configured selective retransmission settings.
|
|
*/
|
|
virtual int SelectiveRetransmissions() const = 0;
|
|
|
|
/*
|
|
* TODO(holmer): Propagate this API to VideoEngine.
|
|
* Sets the selective retransmission settings, which will decide which
|
|
* packets will be retransmitted if NACKed. Settings are constructed by
|
|
* combining the constants in enum RetransmissionMode with bitwise OR.
|
|
* All packets are retransmitted if kRetransmitAllPackets is set, while no
|
|
* packets are retransmitted if kRetransmitOff is set.
|
|
* By default all packets except FEC packets are retransmitted. For VP8
|
|
* with temporal scalability only base layer packets are retransmitted.
|
|
*
|
|
* Returns -1 on failure, otherwise 0.
|
|
*/
|
|
virtual int SetSelectiveRetransmissions(uint8_t settings) = 0;
|
|
|
|
/*
|
|
* Send a Negative acknowledgement packet
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual WebRtc_Word32 SendNACK(const WebRtc_UWord16* nackList,
|
|
const WebRtc_UWord16 size) = 0;
|
|
|
|
/*
|
|
* Store the sent packets, needed to answer to a Negative acknowledgement
|
|
* requests
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual WebRtc_Word32 SetStorePacketsStatus(
|
|
const bool enable,
|
|
const WebRtc_UWord16 numberToStore = 200) = 0;
|
|
|
|
/**************************************************************************
|
|
*
|
|
* Audio
|
|
*
|
|
***************************************************************************/
|
|
|
|
/*
|
|
* set audio packet size, used to determine when it's time to send a DTMF
|
|
* packet in silence (CNG)
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual WebRtc_Word32 SetAudioPacketSize(
|
|
const WebRtc_UWord16 packetSizeSamples) = 0;
|
|
|
|
/*
|
|
* Outband TelephoneEvent(DTMF) detection
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual WebRtc_Word32 SetTelephoneEventStatus(
|
|
const bool enable,
|
|
const bool forwardToDecoder,
|
|
const bool detectEndOfTone = false) = 0;
|
|
|
|
/*
|
|
* Is outband TelephoneEvent(DTMF) turned on/off?
|
|
*/
|
|
virtual bool TelephoneEvent() const = 0;
|
|
|
|
/*
|
|
* Returns true if received DTMF events are forwarded to the decoder using
|
|
* the OnPlayTelephoneEvent callback.
|
|
*/
|
|
virtual bool TelephoneEventForwardToDecoder() const = 0;
|
|
|
|
/*
|
|
* SendTelephoneEventActive
|
|
*
|
|
* return true if we currently send a telephone event and 100 ms after an
|
|
* event is sent used to prevent the telephone event tone to be recorded
|
|
* by the microphone and send inband just after the tone has ended.
|
|
*/
|
|
virtual bool SendTelephoneEventActive(
|
|
WebRtc_Word8& telephoneEvent) const = 0;
|
|
|
|
/*
|
|
* Send a TelephoneEvent tone using RFC 2833 (4733)
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual WebRtc_Word32 SendTelephoneEventOutband(
|
|
const WebRtc_UWord8 key,
|
|
const WebRtc_UWord16 time_ms,
|
|
const WebRtc_UWord8 level) = 0;
|
|
|
|
/*
|
|
* Set payload type for Redundant Audio Data RFC 2198
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual WebRtc_Word32 SetSendREDPayloadType(
|
|
const WebRtc_Word8 payloadType) = 0;
|
|
|
|
/*
|
|
* Get payload type for Redundant Audio Data RFC 2198
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual WebRtc_Word32 SendREDPayloadType(
|
|
WebRtc_Word8& payloadType) const = 0;
|
|
|
|
/*
|
|
* Set status and ID for header-extension-for-audio-level-indication.
|
|
* See http://tools.ietf.org/html/rfc6464 for more details.
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual WebRtc_Word32 SetRTPAudioLevelIndicationStatus(
|
|
const bool enable,
|
|
const WebRtc_UWord8 ID) = 0;
|
|
|
|
/*
|
|
* Get status and ID for header-extension-for-audio-level-indication.
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual WebRtc_Word32 GetRTPAudioLevelIndicationStatus(
|
|
bool& enable,
|
|
WebRtc_UWord8& ID) const = 0;
|
|
|
|
/*
|
|
* Store the audio level in dBov for header-extension-for-audio-level-
|
|
* indication.
|
|
* This API shall be called before transmision of an RTP packet to ensure
|
|
* that the |level| part of the extended RTP header is updated.
|
|
*
|
|
* return -1 on failure else 0.
|
|
*/
|
|
virtual WebRtc_Word32 SetAudioLevel(const WebRtc_UWord8 level_dBov) = 0;
|
|
|
|
/**************************************************************************
|
|
*
|
|
* Video
|
|
*
|
|
***************************************************************************/
|
|
|
|
/*
|
|
* Set the estimated camera delay in MS
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual WebRtc_Word32 SetCameraDelay(const WebRtc_Word32 delayMS) = 0;
|
|
|
|
/*
|
|
* Set the target send bitrate
|
|
*/
|
|
virtual void SetTargetSendBitrate(const WebRtc_UWord32 bitrate) = 0;
|
|
|
|
/*
|
|
* Turn on/off generic FEC
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual WebRtc_Word32 SetGenericFECStatus(
|
|
const bool enable,
|
|
const WebRtc_UWord8 payloadTypeRED,
|
|
const WebRtc_UWord8 payloadTypeFEC) = 0;
|
|
|
|
/*
|
|
* Get generic FEC setting
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual WebRtc_Word32 GenericFECStatus(bool& enable,
|
|
WebRtc_UWord8& payloadTypeRED,
|
|
WebRtc_UWord8& payloadTypeFEC) = 0;
|
|
|
|
|
|
virtual WebRtc_Word32 SetFecParameters(
|
|
const FecProtectionParams* delta_params,
|
|
const FecProtectionParams* key_params) = 0;
|
|
|
|
/*
|
|
* Set method for requestion a new key frame
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual WebRtc_Word32 SetKeyFrameRequestMethod(
|
|
const KeyFrameRequestMethod method) = 0;
|
|
|
|
/*
|
|
* send a request for a keyframe
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual WebRtc_Word32 RequestKeyFrame() = 0;
|
|
};
|
|
} // namespace webrtc
|
|
#endif // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_H_
|